[SCM] Debian packaging for jack-audio-connection-kit branch, master, updated. debian/0.118+svn3796-3-14-g04522ce

Wed Mar 10 20:01:34 UTC 2010

The following commit has been merged in the master branch:
commit 04522ce3ec6af0dd4f182e12111da058a6636d6f
Author: Jonas Smedegaard <dr at jones.dk>
Date:   Wed Mar 10 21:01:01 2010 +0100

    Drop file user-howto and related update rule (7 years old, newest version 4 years old, and seems included in maintained jackd.1 manpage.

diff --git a/debian/jackd.docs b/debian/jackd.docs
index 91f8945..ea42a42 100644
--- a/debian/jackd.docs
+++ b/debian/jackd.docs
@@ -1,4 +1,3 @@
 build-tree/jack-audio-connection-kit-*/TODO
 build-tree/jack-audio-connection-kit-*/AUTHORS
 debian/FAQ
-debian/user-howto
diff --git a/debian/rules b/debian/rules
index bed8164..5268807 100755
--- a/debian/rules
+++ b/debian/rules
@@ -105,11 +105,3 @@ faq:
 	w3m -dump http://jackaudio.org/faq > debian/FAQ.dltmp
 	mv debian/FAQ.dltmp debian/FAQ
 	dch -a "debian/FAQ: updated from webpage"
-
-.PHONY: user-howto
-# this target fetches the user howto from the web
-user-howto:
-	dh_testdir
-	w3m -dump http://www.djcj.org/LAU/jack/ > debian/user-howto.dltmp
-	mv debian/user-howto.dltmp debian/user-howto
-	dch -a "debian/user-howto: updated from webpage"
diff --git a/debian/user-howto b/debian/user-howto
deleted file mode 100644
index 5b749a3..0000000
--- a/debian/user-howto
+++ /dev/null
@@ -1,189 +0,0 @@
-                                                        JACK user documentation
-[jack-logo]
-                                               Last update Tuesday 25 February 2003 14:42
-
-
-What is JACK?
-
-JACK is a low-latency audio server, written primarily for the GNU/Linux
-operating system. It can connect a number of different applications to an audio
-device, as well as allowing them to share audio between themselves. Its clients
-can run in their own processes (ie. as normal applications), or can they can
-run within the JACK server (ie. as a "plugin").
-
-JACK is different from other audio server efforts in that it has been designed
-from the ground up to be suitable for professional audio work. This means that
-it focuses on two key areas: synchronous execution of all clients, and low
-latency operation.
-
-This diagram, using ardour as an example, will give you an overview of how a
-JACKed Linux audio system works.
-
-Jack has two sets of parameter options. The first part are specific to running
-the jack server. The second part are run time options for how jack interfaces
-with the sound driver - currently only ALSA.
-
-The easiest way to start jack is to run this command:
-
-jackd -d alsa -d hw:0
-
-Of course that gives you very little control over what jack does to the audio
-stream and which device you use. You can specify a card name by setting up an
-.asoundrc file. Visit the online ALSA docs for your card/device to get one.
-
-There are many useful options which can be found by typing
-
-jackd -h or jackd -d alsa -h
-
-Example commandlines
-
-Many people with soundblaster live cards find that more appropriate settings
-are:
-
-*jackd -v -d alsa -d (cardnamehere) -p 512
-
-People using RME cards have reported success with:
-
-*jackd -v -d alsa -d (cardnamehere) -p 64
-
-A commandline for starting at 44100hz with verbosity, realtime scheduling,
-hardware monitoring, and shaped dither enabled:
-
-*jackd -v -R -d alsa -d (cardnamehere) -r 44100 -H -z s
-
-You can use ecasound to generate a pure sine wave tone for testing the sound
-quality of your device.
-
-*ecasound -f:32,1,48000 -i null -o jack_alsa,myport -b:1024 -el:sine_fcac,440,1
-
-There are a few other options which you will find useful.
-
-JACK specific options
-
-The default settings for jack are to run at 48000hz with a buffersize of 1024
-frames per second and a period size of 2. Jack currently supports two bitrates.
-Jack's alsa driver/client tries to use SND_PCM_FMT_S32_LE, which is the format
-used by all current 24 bit audio cards except for some USB interfaces that
-actually use 24 bits rather than 24-packed-in-32-bits. If the device can't do
-that, it tries for SND_PCM_FMT_S16_LE, which every audio interface should/does
-support. True 24 bit format wouldn't be a lot of work to support, but its not
-trivial either.
-
-The buffersize determines the latency between when the sound is received by
-jack and when it is sent to the pcm device (the card output). Obviously the
-less the buffersize the more realtime response you will have. Many people have
-found that for general purpose use the default setting is more than adequate
-but when you are doing recording you should set the buffersize as low as your
-card/device can handle without causing sound dropouts (xruns). Some people
-advocate using higher latency for recording to ensure smooth audio. This is a
-tradeoff between realtime response for monitoring and audio quality. It is
-recommended that you test your card and system to find out what the best
-setting is for your setup. 64 frames per interrupt is the lowest currently
-possible in any PC audio hardware. Due to the binary number system you should
-increase the frames in multiples of 2 starting at 64.
-
-For example: 64, 128, 256, 512, 1024, 2048, 4096, 8192....
-
-jackd -v -a -R -P -d
-
--v means verbose. It will output the actions that jack is performing to a
-console. This is very useful for debugging.
--a means to use the inbuilt ASIO support. This can only be enabled on cards
-that support ASIO. ASIO is a protocol developed by Steinburg the makers of many
-Microsoft audio applications. It allows for much lower latency performance
-internal to the soundcard/device.
--R means realtime. This allows you to take full advantage of the low latency
-patches for the Linux kernel. You should enable this if you are doing master
-recordings or want to ensure the applications will receive the audio stream as
-quickly as possible.
--P means Priority. This is superfluous to the -R flag but allows for setting
-the priority of jack to the maximum available. Also useful when you need low
-latency.
--d means driver. This sets the sound driver which jack intefaces with.
-Currently "alsa" is the only option.
-
-Driver specific options
-
-jackd -d alsa -d -r -p -n -H -C -D -C -z
-
-Currently jack only has support for alsa as a sound driver. In the future there
-may be more driver options although it is not very likely.
-
--d means device. This allows you to specify a device other than hw:0
--r means sample rate. Use this to set the number of samples per second that the
-audio is streamed at. 44100Hz is cd quality, 48000Hz (the default) is DAT
-quality, Anything between 44100Hz and 192000Hz is DVD quality. The higher the
-sample rate the more audio data you capture per second and therefore the more
-space you use on your HDD. For many people CD quality is fine. The debate rages
-as to whether sample rates higher than 44100Hz provide better sound quality or
-not. Currently it is at a standoff until someone conducts conclusive double
-blind tests in the tradition of Pepsi vs Coke.
-
-Many people only work at 44100Hz because resampling down from a higher sample
-rate is known to degrade the audio quality when compared to recording at
-44100Hz originally. It is also highly likely that sample libraries you may want
-to use are only available at 44100Hz. Saying that, most people agree that
-acoustic recordings do generally sound better when recorded at higher sample
-rates. Unfortunately CD's are not going to dissapear soon and DVDRW's remain
-expensive so if you want to distribute your recordings it is more than likely
-that they will be shipped at 44100Hz.
-
--p means the frames per period. This is the buffer rate which JACK will stream
-audio at. See above for an explanation of what this means.
--n means periods per hardware buffer. This sets the number of periods per
-interrupt which ALSA polls for your device. Most cards use two periods but some
-use 3, 4 or even 8 or 16 (delta 10/10).
-
-What is the exact purpose of the p and n parameters?
-
-There are several kinds of latency:
-
-    input latency
-    output latency
-    through (or "roundtrip") latency
-
-    p affects input latency: how long from when a piece of data arrives at the
-    audio interface connectors until user space software can use it?
-
-    p*n affects output latency: how long from when a piece of data is delivered
-    by user space data until it leaves the audio interface connectors?
-
-    Roundtrip latency is combination of these two.
-
-Conventional low latency systems (e.g. ASIO) use n=2 all the time. ALSA is
-rather unusual in allowing other values.
-
--H means Hardware monitoring. This is only available with cards/devices that
-support this feature. Usually cards that support ASIO will support hardware
-monitoring. It allows you to hear the audio stream flowing through the pcm in/
-outs at that very moment. This is very good for hearing what you are recording
-as you are recording it.
--C means capture only. This opens the ALSA driver in read only mode which is
-useful for people who only want to record audio and don't have a need to hear
-what they are recording.
--D means duplex. This opens the ALSA driver in read/write mode which means that
-you can play and record at the same time. Most people will only want to use
-this which is the default mode anyway.
--P means playback only. This opens the ALSA driver in write only mode which is
-useful for people who have no inputs or only want to play audio not record. It
-can also reduce latency.
-
--z means dither. There are currently four options to the dither flag.
--z r means rectangular dither.
--z t means triangular dither.
--z s means shaped dither.
--z - means no dither(the default).
-
-Dither is used to make the audio cleaner. The best way to describe it is to
-imagine a painting with many dots. If you view it up close you can see each dot
-and the image is not very clear. If you view it from far away the image becomes
-clearer because your eyes/brain dither the dots to smooth out the image. It is
-a murky subject and obviously a very personal choice as to what dither is the
-best. For most people it is just plain magic. Anyone running at 16bit who cares
-about quality or has CPU cycles to spare should run with dither. Triangular is
-probably the best compromise of quality vs cpu cost (its very fast), but shaped
-is the best.
-
-Document prepared by Patrick Shirkey <pshirkey_at_boosthardware.com>
-Thanks to everyone who contributes, wittingly or not...
-

-- 
Debian packaging for jack-audio-connection-kit

